5783 Commits (855746fb75d5852eb4ee2bd4c99b2debbfa045e0)

Author	SHA1	Message	Date
Joshua Colp	ddae0bab23	Merge "tcptls: Change error message to debug." into 15	9 years ago
Joshua Colp	7fa11842d8	tcptls: Change error message to debug. ... The Websocket implementation will steal the underlying stream of TCP/TLS sessions. This results in an error message being output about a stream not being present when in reality this is actually fine. This change moves it to a debug message instead. Change-Id: I66cc639080b4b4599beadb4faa7d313f2721d094	9 years ago
George Joseph	752f0533dd	res_pjsip: Add handling for incoming unsolicited MWI NOTIFY ... A new endpoint parameter "incoming_mwi_mailbox" allows Asterisk to receive unsolicited MWI NOTIFY requests and make them available to other modules via the stasis message bus. res_pjsip_pubsub has a new handler "pubsub_on_rx_mwi_notify_request" that parses a simple-message-summary body and, if endpoint->incoming_mwi_account is set, calls ast_publish_mwi_state with the voice-message counts from the message. Change-Id: I08bae3d16e77af48fcccc2c936acce8fc0ef0f3c	9 years ago
George Joseph	a824fcc8e8	stasis/control: Fix possible deadlock with swap channel ... If an error occurs during a bridge impart it's possible that the "bridge_after" callback might try to run before control_swap_channel_in_bridge has been signalled to continue. Since control_swap_channel_in_bridge is holding the control lock and the callback needs it, a deadlock will occur. * control_swap_channel_in_bridge now only holds the control lock while it's actually modifying the control structure and releases it while the bridge impart is running. * bridge_after_cb is now tolerant of impart failures. Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3	9 years ago
Walter Doekes	329e111947	res/res_pjsip: Standardize/fix localnet checks across pjsip. ... In 2dee95cc (ASTERISK-27024) and 776ffd77 (ASTERISK-26879) there was confusion about whether the transport_state->localnet ACL has ALLOW or DENY semantics. For the record: the localnet has DENY semantics, meaning that "not in the list" means ALLOW, and the local nets are in the list. Therefore, checks like this look wrong, but are right: /* See if where we are sending this request is local or not, and if not that we can get a Contact URI to modify */ if (ast_apply_ha(transport_state->localnet, &addr) != AST_SENSE_ALLOW) { ast_debug(5, "Request is being sent to local address, " "skipping NAT manipulation\n"); (In the list == localnet == DENY == skip NAT manipulation.) And conversely, other checks that looked right, were wrong. This change adds two macro's to reduce the confusion and uses those instead: ast_sip_transport_is_nonlocal(transport_state, addr) ast_sip_transport_is_local(transport_state, addr) ASTERISK-27248 #close Change-Id: Ie7767519eb5a822c4848e531a53c0fd054fae934	9 years ago
Sean Bright	1d33757aa0	rtp_engine: Prevent possible double free with DTLS config ... ASTERISK-27225 #close Reported by: Richard Kenner Change-Id: I097b81734ef730f8603c0b972909d212a3a5cf89	9 years ago
Corey Farrell	0372157a48	AST-2017-006: Fix app_minivm application MinivmNotify command injection ... An admin can configure app_minivm with an externnotify program to be run when a voicemail is received. The app_minivm application MinivmNotify uses ast_safe_system() for this purpose which is vulnerable to command injection since the Caller-ID name and number values given to externnotify can come from an external untrusted source. * Add ast_safe_execvp() function. This gives modules the ability to run external commands with greater safety compared to ast_safe_system(). Specifically when some parameters are filled by untrusted sources the new function does not allow malicious input to break argument encoding. This may be of particular concern where CALLERID(name) or CALLERID(num) may be used as a parameter to a script run by ast_safe_system() which could potentially allow arbitrary command execution. * Changed app_minivm.c:run_externnotify() to use the new ast_safe_execvp() instead of ast_safe_system() to avoid command injection. * Document code injection potential from untrusted data sources for other shell commands that are under user control. ASTERISK-27103 Change-Id: I7552472247a84cde24e1358aaf64af160107aef1	9 years ago
Jenkins2	d05ab9e91a	Merge "core: Reduce video update queueing." into 15	9 years ago
Joshua Colp	6baf2c26f3	core: Reduce video update queueing. ... A video update frame is used to indicate that a channel with video negotiated should provide a full frame so the decoder decoding the stream is able to do so. In situations where a queue is used to store frames it makes no sense for the queue to contain multiple video update frames. One is sufficient to have a full frame be sent. ASTERISK-27222 Change-Id: Id3f40a6f51b740ae4704003a1800185c0c658ee7	9 years ago
Richard Mudgett	3e9cd53b5c	bridge_channel.c: Fix FRACK when mapping frames to the bridge. ... * Add protection checks when mapping streams to the bridge. The channel and bridge may be in the process of updating the stream mapping when a media frame comes in so we may not be able to map the frame at the time. * We need to map the streams to the bridge's stream numbers right before they are written into the bridge. That way we don't have to keep locking/unlocking the bridge and we won't have any synchronization problems before the frames actually go into the bridge. * Protect the deferred queue with the bridge_channel lock. ASTERISK-27212 Change-Id: Id6860dd61b594b90c8395f6e2c0150219094c21a	9 years ago
Richard Mudgett	efa6262814	bridge: Fix softmix bridge deadlock. ... * Fix deadlock in bridge_softmix.c:softmix_bridge_stream_topology_changed() between bridge_channel and channel locks. * The new bridge technology topology change callbacks must be called with the bridge locked. The callback references the bridge channel list, the bridge technology could change, and the bridge stream mapping is updated. ASTERISK-27212 Change-Id: Ide4360ab853607e738ad471721af3f561ddd83be	9 years ago
Richard Mudgett	6c555891eb	channel: Fix topology API locking. ... * ast_channel_request_stream_topology_change() must not be called with any channel locks held. * ast_channel_stream_topology_changed() must be called with only the passed channel lock held. ASTERISK-27212 Change-Id: I843de7956d9f1cc7cc02025aea3463d8fe19c691	9 years ago
Kevin Harwell	f919adac8d	manager: hook event is not being raised ... When the iostream code went in it introduced a conditional that made it so the hook event was not being raised even if a hook is present. This patch adds a check to see if a hook is present in astman_append. If so then call into the send_string function, which in turn raises the even for specified hook. Also updated the ami hooks unit test, so the test could be automated. ASTERISK-27200 #close Change-Id: Iff37f02f9708195d8f23e68f959d6eab720e1e36	9 years ago
Richard Mudgett	89a92b1f75	STUN/netsock2: Fix some valgrind uninitialized memory findings. ... * netsock2.c: Test the addr->len member first as it may be the only member initialized in the struct. * stun.c:ast_stun_handle_packet(): The combinded[] local array could get used uninitialized by ast_stun_request(). The uninitialized string gets copied to another location and could overflow the destination memory buffer. These valgrind findings were found for ASTERISK_27150 but are not necessarily a fix for the issue. Change-Id: I55f8687ba4ffc0f69578fd850af006a56cbc9a57	9 years ago
Jenkins2	b24854e16b	Merge "bridge: Fix stream topology/participant locking and video misrouting." into 15	9 years ago
Joshua Colp	0e352ec510	bridge: Fix stream topology/participant locking and video misrouting. ... This change fixes a few locking issues and some video misrouting. 1. When accessing the stream topology of a channel the channel lock must be held to guarantee the topology remains valid. 2. When a channel was joined to a bridge the bridge specific implementation for stream mapping was not invoked, causing video to be misrouted for a brief period of time. ASTERISK-27182 Change-Id: I5d2f779248b84d41c5bb3896bf22ba324b336b03	9 years ago
Corey Farrell	3dbb1b9f48	channel: Fix leak on successful call to chan->tech->requester. ... joint_cap needs to be released unconditionally as chan->tech->requester does not steal the reference even on success. ASTERISK-27180 #close Change-Id: I647728992559bdb0a9c7357c20be1b36400d68b6	9 years ago
Corey Farrell	cd79a15b2f	Fix compile error for old versions of GCC. ... Use -Wno-format-truncation only if supported by compiler. ASTERISK-27171 #close Change-Id: Iac0aed7a5bcaa16c21b7d62c4e4678d244c4ccb6	9 years ago
Corey Farrell	7c3cdc81f7	Fix compiler warnings on Fedora 26 / GCC 7. ... GCC 7 has added capability to produce warnings, this fixes most of those warnings. The specific warnings are disabled in a few places: * app_voicemail.c: truncation of paths more than 4096 chars in many places. * chan_mgcp.c: callid truncated to 80 chars. * cdr.c: two userfields are combined to cdr copy, fix would break ABI. * tcptls.c: ignore use of deprecated method SSLv3_client_method(). ASTERISK-27156 #close Change-Id: I65f280e7d3cfad279d16f41823a4d6fddcbc4c88	9 years ago
Joshua Colp	1c07c80eef	Merge "bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues." into 15	9 years ago
Jenkins2	fab609deeb	Merge "Core: Add support for systemd socket activation." into 15	9 years ago
Joshua Colp	451d86d62e	bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues. ... This change does a few things to improve packet loss and renegotiation: 1. On outgoing RTP streams we will now properly reflect out of order packets and packet loss in the sequence number. This allows the remote jitterbuffer to better reorder things. 2. Video updates can now be discarded for a period of time after one has been sent to prevent flooding of clients. 3. For declined and removed streams we will now release any media session resources associated with them. This was not previously done and caused an issue where old state was being used for a new stream. 4. RTP bundling was not actually removing bundled RTP instances from the parent. This has been resolved by removing based on the RTP instance itself and not the SSRC. 5. The code did not properly handle explicitly unbundling an RTP instance from its parent. This now works as expected. ASTERISK-27143 Change-Id: Ibd91362f0e4990b6129638e712bc8adf0899fd45	9 years ago
Jenkins2	29a746c534	Merge "core: Add VP9 passthrough support." into 15	9 years ago
Joshua Colp	927fc6bbd9	core: Add VP9 passthrough support. ... This change adds VP9 as a known codec and creates a cached "vp9" media format for use. Change-Id: I025a93ed05cf96153d66f36db1839109cc24c5cc	9 years ago
Rusty Newton	715d79b60d	say.c: Fix file locations for second, seconds, minute, minutes files ... The seconds and minutes files have always existed in the base language directory of the Core package. So say.c has always been calling the wrong location (under digits/) for those two files and in the case of second and minute they didn't exist in the Core packages at all. The 1.6 sounds release moves the second and minute files into Core from Extra for the languages that already had them. A future release will include the second and minute files for languages that didn't already have them. This patch just changes all the target locations for second, seconds, minute, and minutes that were under the digits subdir to be under the root of sounds instead. Which is where the sounds will be for some languages after 1.6 sounds and for all languages after a future release. ASTERISK-25810 #close Change-Id: I05d9d4bee6a7237030530a46e7eb3df15f13f702 Reported-by: Nicolas Riendeau	9 years ago
Corey Farrell	eea9da2f42	Core: Add support for systemd socket activation. ... This change adds support for socket activation of certain SOCK_STREAM listeners in Asterisk: * AMI / AMI over TLS * CLI * HTTP / HTTPS Example systemd units are provided. This support extends to any socket which is initialized using ast_tcptls_server_start, so any unknown modules using this function will support socket activation. Asterisk continues to function as normal if socket activation is not enabled or if systemd development headers are not available during build. ASTERISK-27063 #close Change-Id: Id814ee6a892f4b80d018365c8ad8d89063474f4d	9 years ago
Joshua Colp	f43fc91911	Merge "core: Add digit filtering to ast_waitfordigit_full"	9 years ago
Jenkins2	62c381afdb	Merge "core: Add PARSE_TIMELEN support to ast_parse_arg and ACO."	9 years ago
Jenkins2	594c7a50af	Merge "bridge/core_unreal: Fix SFU bugs with forwarding frames."	9 years ago
Jenkins2	0f45c979a3	Merge "res_rtp_asterisk / res_pjsip: Add support for BUNDLE."	9 years ago
Corey Farrell	78a50b0343	core: Add PARSE_TIMELEN support to ast_parse_arg and ACO. ... This adds support for parsing timelen values from config files. This includes support for all flags which apply to PARSE_INT32. Support for this parser is added to ACO via the OPT_TIMELEN_T option type. Fixes an issue where extra characters provided to ast_app_parse_timelen were ignored, they now cause an error. Testing is included. ASTERISK-27117 #close Change-Id: I6b333feca7e3f83b4ef5bf2636fc0fd613742554	9 years ago
Joshua Colp	065c3005ad	res_rtp_asterisk / res_pjsip: Add support for BUNDLE. ... BUNDLE is a specification used in WebRTC to allow multiple streams to use the same underlying transport. This reduces the number of ICE and DTLS negotiations that has to occur to 1 normally. This change implements this by adding support for it to the RTP SDP module in PJSIP. BUNDLE can be turned on using the "bundle" option and on an offer we will offer to bundle streams together. On an answer we will accept any bundle groups provided. Once accepted each stream is bundled to another RTP instance for transport. For the res_rtp_asterisk changes the ability to bundle an RTP instance to another based on the SSRC received from the remote side has been added. For outgoing traffic if an RTP instance is bundled to another we will use the other RTP instance for any transport related things. For incoming traffic received from the transport instance we look up the correct instance based on the SSRC and use it for any non-transport related data. ASTERISK-27118 Change-Id: I96c0920b9f9aca7382256484765a239017973c11	9 years ago
Corey Farrell	6b138046e7	core: Add digit filtering to ast_waitfordigit_full ... This adds a parameter to ast_waitfordigit_full which can be used to only stop waiting when certain expected digits are received. Any unexpected DTMF digits are simply ignored. This also creates a new dialplan application WaitDigit. ASTERISK-27129 #close Change-Id: Id233935ea3d13e71c75a0861834c5936c3700ef9	9 years ago
Joshua Colp	95b35cb1cb	Merge "Avoid setting maxfiles for a remote asterisk"	9 years ago
Jenkins2	fbcfa6b4b2	Merge "http.c: Reduce log spam"	9 years ago
Joshua Colp	7f09fd2c2f	bridge/core_unreal: Fix SFU bugs with forwarding frames. ... This change fixes a few things uncovered during SFU testing. 1. Unreal channels incorrectly forwarded video frames when no video stream was present on them. This caused a crash when they were read as the core requires a stream to exist for the underlying media type. The Unreal channel will now ensure a stream exists for the media type before forwarding the frame and if no stream exists then the frame is dropped. 2. Mapping of frames during bridging from the stream number of the underlying channel to the stream number of the bridge was done in the wrong location. This resulted in the frame getting dropped. This mapping now occurs on reading of the frame from the channel. 3. Bridging was using the wrong ast_read function resulting in it living in a non-multistream world. 4. In bridge_softmix when adding new streams to existing channels the wrong stream topology was copied resulting in no streams being added. Change-Id: Ib7445722c3219951d6740802a0feddf2908c18c8	9 years ago
Tzafrir Cohen	d58ef31acd	Avoid setting maxfiles for a remote asterisk ... Setting maxfiles (maximum number of open files) has no practical effect on a remote asterisk (rasterisk, rasterisk -x). It has an ill effect of printing an extra message, which may be annoying in case of -x. ASTERISK-27105 #close Change-Id: Iaf9eb344e4b4b517df91b736b27ec55f6a6921a2	9 years ago
George Joseph	303f935a50	http.c: Reduce log spam ... Messages like "fwrite() failed: Connection reset by peer" are no help whatsoever, especially since they can be caused simply by a client disconnecting. * Make those WARNINGs DEBUGs. * Check the return from ast_iostream_printf of headers. Change-Id: I17bd5f3621514152a7b2b263c801324c5e96568b	9 years ago
Richard Mudgett	03ae8b0105	json.c: Add backtrace log to find 'Invalid UTF-8 string' errors ... Change-Id: I9020ff9f2b3749904317c0c173f47a1bbed6f929	9 years ago
Jenkins2	d6c08cc559	Merge "core: Remove 'Data Retrieval API'"	9 years ago
Sean Bright	325eeced6a	core: Remove 'Data Retrieval API' ... This API was not actively maintained, was not added to new modules (such as res_pjsip), and there exist better alternatives to acquire the same information, such as the ARI. Change-Id: I4b2185a83aeb74798b4ad43ff8f89f971096aa83	9 years ago
Corey Farrell	50ddb56dad	channel: Clear channel flag in error branch. ... Clear channel flag AST_FLAG_END_DTMF_ONLY in ast_waitfordigit_full when ast_read returns NULL. ASTERISK-27100 #close Change-Id: Id3039e9a4e74e0cb359f636c9fd0c9740ebf7d9d	9 years ago
Mark Michelson	45df25a579	chan_pjsip: Add support for multiple streams of the same type. ... The stream topology (list of streams and order) is now stored with the configured PJSIP endpoints and used during the negotiation process. Media negotiation state information has been changed to be stored in a separate object. Two of these objects exist at any one time on a session. The active media state information is what was previously negotiated and the pending media state information is what the media state will become if negotiation succeeds. Streams and other state information is stored in this object using the index (or position) of each individual stream for easy lookup. The ability for a media type handler to specify a callback for writing has been added as well as the ability to add file descriptors with a callback which is invoked when data is available to be read on them. This allows media logic to live outside of the chan_pjsip module. Direct media has been changed so that only the first audio and video stream are directly connected. In the future once the RTP engine glue API has been updated to know about streams each individual stream can be directly connected as appropriate. Media negotiation itself will currently answer all the provided streams on an offer within configured limits and on an offer will use the topology created as a result of the disallow/allow codec lines. If a stream has been removed or declined we will now mark it as such within the resulting SDP. Applications can now also request that the stream topology change. If we are told to do so we will limit any provided formats to the ones configured on the endpoint and send a re-invite with the new topology. Two new configuration options have also been added to PJSIP endpoints: max_audio_streams: determines the maximum number of audio streams to offer/accept from an endpoint. Defaults to 1. max_video_streams: determines the maximum number of video streams to offer/accept from an endpoint. Defaults to 1. ASTERISK-27076 Change-Id: I8afd8dd2eb538806a39b887af0abd046266e14c7	9 years ago
Jenkins2	01536546e2	Merge "bridge: stuck channel(s) after failed attended transfer"	9 years ago
Kevin Harwell	27dae55fb6	core_local: local channel data not being properly unref'ed and unlocked ... In an earlier version of Asterisk a local channel [un]lock all functions were added in order to keep a crash from occurring when a channel hung up too early during an attended transfer. Unfortunately, when a transfer failure occurs and depending on the timing, the local channels sometime do not get properly unlocked and deref'ed after being locked and ref'ed. This happens because the underlying local channel structure gets NULLed out before unlocking. This patch reworks those [un]lock functions and makes sure the values that get locked and ref'ed later get unlocked and deref'ed. ASTERISK-27074 #close Change-Id: Ice96653e29bd9d6674ed5f95feb6b448ab148b09	9 years ago
Kevin Harwell	45a1f4e2ae	bridge: stuck channel(s) after failed attended transfer ... If an attended transfer failed it was possible for some of the channels involved to get "stuck" because Asterisk was not hanging up the transfer target. This patch ensures Asterisk hangs up the transfer target when an attended transfer failure occurs. ASTERISK-27075 #close Change-Id: I98a6ecd92d3461ab98c36f0d9451d23adaf3e5f9	9 years ago
Joshua Colp	57bbba7d43	Merge "res_stasis: Plug reference leak on stolen channels"	9 years ago
George Joseph	3f5bf287a2	Merge "SDP: Add get/set option calls for RTP sched context per type."	9 years ago
George Joseph	854a6de819	res_stasis: Plug reference leak on stolen channels ... When a stasis channel is stolen by another app, the control structure is unreffed but never unlinked from the app_controls container. This causes the channel reference to leak. Added OBJ_UNLINK to the callback in channel_stolen_cb. Also added some additional channel lifecycle debug messages to channel.c. ASTERISK-27059 #close Repoorted-by: George Joseph Change-Id: Ib820936cd49453f20156971785e7f4f182c56e14	9 years ago
Frederic LE FOLL	0ad95bc8a0	Core/PBX: Deadlock between dialplan execution and application unregistration. ... Not easy to reproduce, but we have noticed deadlocks when unloading a module while dialplan is handling a request. The deadlock is between : 1) Dialplan execution: pbx_extension_helper() first taking conlock, then pbx_findapp() [when called] asking for lock on apps list. 2) Application unregistration: ast_unregister_application() first taking lock on apps list, then unreference_cached_app() [when called] asking for conlock. As a protection, I suggest to modify ast_unregister_application(), so that it anticipates the need of conlock, before taking the lock on apps list. The side effect is a longer unavailability of conlock when unregistering an application. ASTERISK-27041 Change-Id: I0db0f1eb320da6a5758cce3a47d765be1face8e2	9 years ago